Reply by Jerry Avins May 21, 20052005-05-21
ahmad wrote:
> sir i have mailed u at your address, plz offer me to remain intact with > you so that i soon learn my DSP initials. thanx
My response: ahmad wrote: > Sir i am a fresh graduate of computer science, and a kid in the field > of DSP. i want your continuous cooperation. Please reply me with the > answer "you are always welcome" thanx for answering this "Indeed, you are always welcome to ask questions, but you should ask them in comp.dsp and not in my personal email. You get the benefit of many people's knowledge and experience; mine alone is much more limited. I am often away, so you may also get quicker response from there. Jerry -- When you know how things work, the world is one big sandbox. ������������������������������������������������������������" Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ahmad May 21, 20052005-05-21
sir i have mailed u at your address, plz offer me to remain intact with
you so that i soon learn my DSP initials. thanx

Reply by ahmad May 21, 20052005-05-21
sir i have mailed u at your address, plz offer me to remain intact with
you so that i soon learn my DSP initials. thanx

Reply by Don May 21, 20052005-05-21
point1.

No thats not a good idea to multiply and divive the samples values to
achive the increase and decrease the volume of the audio stream.

Point 2.

If you have to still want to go ahead with this then i will suggest
that keep the tarck for the sampling values (Which you multiply
/ divide) so that it may not get over flow.

Lets Say example :
you are implementing for 16 bit platform.

  S1 = suppose you have value 32756
  and you multiply with 2 ..... the there will be
  overflow, And you will get the audio stram
  distorted ... same is the case for dividing

  So the key over here is to keep for the track
  of the sampled value which you multiply and
  divide.

point 3:

   BTW, why you want to get Fast play and slow
   play ??, I suppose you dont need this if it is
   for the real time application.

Point 4:
   First Under satnd the Concept of pitch.
   once you are done then you can get the
   Bass and Trebblle.

One more thing you are implementing for
Music File or Speech File.

Reagrds
Don:

Reply by ahmad May 20, 20052005-05-20
sir i am extremely grateful to you for spending sometime for me and
providing me your knowledge.... Thanx again

Reply by Jerry Avins May 20, 20052005-05-20
ahmad wrote:
> sir i want to make a wave player and want to implement such functions. > 1. increasing and decreasing volume ( i have some idea to implement it > that multiply/divide each sample with a constant ) > 2. Fast play and slow play while stream is playing > 3. increasing/decreasing pitch of sound > 4. equilizer ( how can i increase bass and treble ) > > i want some tips to do this all plz send me your suggestions > waiting for ur reply > with regards
That may be a small package, but it is an enormous undertaking. Read the applicable articles on http://dspdimension.com/ to understand pitch changing, (more at http://www.mega-nerd.com/SRC/ and wave file formats at http://www.mega-nerd.com/libsndfile/. There are tutorials at http://www.dspguru.com/, http://www.mega-nerd.com/Res/IADSPL/, and elsewhere. Those should give you an idea of the scope of what you want to do. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ahmad May 20, 20052005-05-20
sir i want to make a wave player and want to implement such functions.
1. increasing and decreasing volume ( i have some idea to implement it
that multiply/divide each sample with a constant )
2. Fast play and slow play while stream is playing
3. increasing/decreasing pitch of sound
4. equilizer ( how can i increase bass and treble )

i want some tips to do this all plz send me your suggestions
waiting for ur reply
with regards